similar to: How do we updated to the new .7.1 version.

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

Referring to my previous post about degradation of voice quality when having more than 2 connection. The actual route is: pc xlite -> local asterisk box -> iaxtel -> local asterisk I have tried out a different situation: pc xlite -> local asterisk box -> iaxtel and the second connection pc xlite -> local asterisk box -> iaxtel -> local asterisk The same degradation

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/a6f17a16/attachment.htm -------------- next part -------------- Hello I found related question on vmail.cgi in the mailing list but that didn't answer my question. I did copy the vmail.cgi to /var/www/cgi-bin/ but still gets the following error message when I access

sorry for the cross post, but this is germane to the developpers as well as the larger user community. Re: [SIP 0000104]: [patch] Cisco-like NAT trick for outbound SIP connections On Sat, Jan 03, 2004 at 08:07:29PM -0600, bugs@digium.com wrote: > > A BUGNOTE has been added to this bug. tabarnac! it's been months now! the only thing that i can think at this point is that mark

> But this is not to say _you_ can't built a reliable VOIP based > system. Get _two_ providers and set up your dial plan in > extensions.conf to "fail over" if one service fails to > connect to dial via the next one and finally if both fail > use pstn. your users will see a system the "just works". Now there's an idea. I'm playing with this now,

hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (? 216) plus a small ?50 drive and cables, so say ?300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP

Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

thanks for the reply. the problem is after i start icecast with icecast -c /path to icecast.xml, i see the following: Changed groupid to 501. Changed userid to 501. and then it just hangs. i do not get a command prompt or anything. so i was wondering if there are any other messages that normally follow during the startup process. <p>--- Karl Heyes <karl@xiph.org> wrote: > On

Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks

Hello! Has anyone gotten 4.9R to recognize the "ServerWorks CSB6" ATA controller on a Dell 600SC? How about installing 5.2R on a 600SC? 5.2R recognizes the said controller, but it corrupts data. -- eof

Hello Everyone, I'm using Mandrake 10 and trying to learng Samba 3.0.2a. I compile the source and install it alright on my Mandrake linux (./configure, make, and make install). Here what I'd done after the installation: Add to /etc/services file swat 901/tcp Add to /etc/inetd.conf file swat stream tcp nowait.400 root /usr/local/samba/sbin swat Since swat binary in the

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hi, I have a problem of loss of file ownership with rsync. For some odd reasons, I have to use a windows machine as rsync server, and to backup/restore a filesystem from/to a linux client. All files in my linux filesystem are owned by root, with gid root. If I save the files onto the windows rsync server from linux using rsync, and then restore them back to linux, the ownership for all files

Hello, What do you recommend for keeping track of user activities? For preserving bash histories I followed these recommendations: http://www.defcon1.org/secure-command.html They include using 'chflags sappnd .bash_history', enabling process accounting, and the like. My goal is to "watch the watchers," i.e. watch for abuse of power by SOC people with the ability to view

Hi, I'm running a simple test from asterisk towards a public telco switch (AXE10) over E100P. Here is the test case: 1) 30 calls are setup simultainously, 20 sec ringing time. 2) no calls answers (just calling a vacant public tax office :=) 3) Each channel will continue on its own with the same proceedure: ring for 20 sec, then hangup, ring for 20 sec, and so on. Of course this leads to

i have downloaded .flac files. can anyone tell me how to convert these files to .wav or .mp3? pppppppppppllllllllllllllllllleeeeeeeeeeeeeeaaaaaaaaaaaaaaaassssssssssssssseeeeeeee?????!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20040111/fc527cea/attachment.htm

On Wed, 7 Jan 2004, jensen galan wrote: > <hostname>localhost</hostname> > <port>8000</port> > > <listen-socket> > <port>8000</port> > <bind-address>192.168.1.99</bind-address> > </listen-socket> ah. I think I see your problem. You don't want to use both of these port values. If you want it to just use